Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.
This CL was generated by the following script:
git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
git cl format
Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.
Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index bb7f890..3528314 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -917,6 +917,7 @@
"../../rtc_base:rtc_numerics",
"../../rtc_base:safe_minmax",
"../../system_wrappers:field_trial_api",
+ "//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
public_deps = [
@@ -1034,6 +1035,7 @@
"../../rtc_base/system:file_wrapper",
"../../system_wrappers",
"../../system_wrappers:field_trial_api",
+ "//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
@@ -1775,6 +1777,7 @@
testonly = true
deps = audio_coding_deps + [
+ "//third_party/abseil-cpp/absl/memory",
"../..:typedefs",
":audio_coding",
":neteq_input_audio_tools",
@@ -2248,6 +2251,7 @@
"../../test:test_common",
"../../test:test_support",
"//testing/gtest",
+ "//third_party/abseil-cpp/absl/memory",
]
defines = audio_coding_defines
diff --git a/modules/audio_coding/audio_network_adaptor/event_log_writer.cc b/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
index 9cdbc54..4a92343 100644
--- a/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
+++ b/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
@@ -12,10 +12,10 @@
#include <algorithm>
+#include "absl/memory/memory.h"
#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
-#include "rtc_base/ptr_util.h"
namespace webrtc {
@@ -63,9 +63,9 @@
}
void EventLogWriter::LogEncoderConfig(const AudioEncoderRuntimeConfig& config) {
- auto config_copy = rtc::MakeUnique<AudioEncoderRuntimeConfig>(config);
- event_log_->Log(
- rtc::MakeUnique<RtcEventAudioNetworkAdaptation>(std::move(config_copy)));
+ auto config_copy = absl::make_unique<AudioEncoderRuntimeConfig>(config);
+ event_log_->Log(absl::make_unique<RtcEventAudioNetworkAdaptation>(
+ std::move(config_copy)));
last_logged_config_ = config;
}
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 05d3b72..e6240e6 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -14,6 +14,7 @@
#include <iterator>
#include <utility>
+#include "absl/memory/memory.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
@@ -25,7 +26,6 @@
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/protobuf_utils.h"
-#include "rtc_base/ptr_util.h"
#include "rtc_base/string_to_number.h"
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/field_trial.h"
@@ -239,7 +239,7 @@
const AudioEncoderOpusConfig& config,
int payload_type) {
RTC_DCHECK(config.IsOk());
- return rtc::MakeUnique<AudioEncoderOpusImpl>(config, payload_type);
+ return absl::make_unique<AudioEncoderOpusImpl>(config, payload_type);
}
absl::optional<AudioCodecInfo> AudioEncoderOpusImpl::QueryAudioEncoder(
@@ -388,7 +388,7 @@
return DefaultAudioNetworkAdaptorCreator(config_string, event_log);
},
// We choose 5sec as initial time constant due to empirical data.
- rtc::MakeUnique<SmoothingFilterImpl>(5000)) {}
+ absl::make_unique<SmoothingFilterImpl>(5000)) {}
AudioEncoderOpusImpl::AudioEncoderOpusImpl(
const AudioEncoderOpusConfig& config,
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index dde2090..7a6d5fd 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -12,6 +12,7 @@
#include <memory>
#include <utility>
+#include "absl/memory/memory.h"
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "common_audio/mocks/mock_smoothing_filter.h"
#include "common_types.h" // NOLINT(build/include)
@@ -21,7 +22,6 @@
#include "modules/audio_coding/neteq/tools/audio_loop.h"
#include "rtc_base/checks.h"
#include "rtc_base/fakeclock.h"
-#include "rtc_base/ptr_util.h"
#include "test/field_trial.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -64,7 +64,7 @@
std::unique_ptr<AudioEncoderOpusStates> CreateCodec(size_t num_channels) {
std::unique_ptr<AudioEncoderOpusStates> states =
- rtc::MakeUnique<AudioEncoderOpusStates>();
+ absl::make_unique<AudioEncoderOpusStates>();
states->mock_audio_network_adaptor = nullptr;
states->fake_clock.reset(new rtc::ScopedFakeClock());
states->fake_clock->SetTimeMicros(kInitialTimeUs);
diff --git a/modules/audio_coding/neteq/tools/rtp_encode.cc b/modules/audio_coding/neteq/tools/rtp_encode.cc
index 1984e3f..abbb621 100644
--- a/modules/audio_coding/neteq/tools/rtp_encode.cc
+++ b/modules/audio_coding/neteq/tools/rtp_encode.cc
@@ -21,6 +21,7 @@
#include <map>
#include <string>
+#include "absl/memory/memory.h"
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/L16/audio_encoder_L16.h"
#include "api/audio_codecs/g711/audio_encoder_g711.h"
@@ -33,7 +34,6 @@
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "rtc_base/flags.h"
#include "rtc_base/numerics/safe_conversions.h"
-#include "rtc_base/ptr_util.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
@@ -313,7 +313,7 @@
AudioEncoderCng::Config cng_config = GetCngConfig(codec->SampleRateHz());
RTC_DCHECK(codec);
cng_config.speech_encoder = std::move(codec);
- codec = rtc::MakeUnique<AudioEncoderCng>(std::move(cng_config));
+ codec = absl::make_unique<AudioEncoderCng>(std::move(cng_config));
}
RTC_DCHECK(codec);